1. Field of the Invention
The function addressed is that of adaptive equalization to mitigate the effects of multiple path radio propagation within the below stated context. Without this equalization, fading and intersymbol interference degrades the accuracy of the recovered bit stream. This degradation of the received signal is caused by the vector sum of multiple copies of the transmitted signal arriving with various time offsets and amplitudes,
Prior art methods are only economically applicable to point-to-point links. Such prior art methods applied to point-to-multipoint, two-way links are not only uneconomic but consume excessive channel time.
The context of this invention is radio-linked, high-data-transfer-rate communication at microwave frequencies. More specifically, the context is in the use of the radio path for communication between one fixed access point common to and serving multiple variously located user stations. The context is limited to the further subset in which time division duplex is used alternately to transmit up and down link information on a single radio channel. In this context, the radio path is identical in either direction when the same antenna is used for both transmitting and receiving; and the communication of each station is time division multiplexed on the common radio channel.
2. Description of the Prior Art
Prior art time division multiplexing requires that each station transmit in short bursts preceded by a preamble from which the receiver must condition itself for the body of the message. Time division may be with regular slots as is the case for virtually all telecom applications, or at irregular intervals as practiced in computer local area networks transferring packets. If a system must perform both of these functions, it is convenient and advantageous to have regularly periodic frames tied to the public network timing function. A reasonable choice of frame period is equivalent to 48 octets (8-bits) every 6 milliseconds to provide a virtual circuit of 64 Kbps. For voice circuits requiring only 32 Kbps, an alternative is to use only every other frame. The number 48 is the octet payload size of one cell in ATM (asynchronous transfer mode) communication.
Equalization Function
Prior art equalization occurs in the developing European Local Area Network Standard known as HIPERLAN (High Performance Local Area Network) prepared by the ETSI RES 10 Committee. In this system, each transmitted burst is at rate of 25 Mbps and begins with a "training symbol" of 480 bits that enables the destination receivers to infer the compensation necessary from the difference between the actual received signal and its known original pattern. Considerable system channel time is required for this function because it is required on every transmission regardless of its size.
From limitations on access and transfer delay across the system, it is undesirable to have long bursts. For telecom, the quantizing delay is the period of time required for the necessary number of octets to be accumulated in a buffer. For good transmission reasons, it is undesirable for this delay to exceed 12 milliseconds. With voice bandwidth compression, this number is commonly exceeded in many telephone systems, and which then require echo cancellation circuits.
One ATM cell is 53 octets or 424 bits. Several additional octets must be added for radio system overhead and acquisition preamble. It may be seen that the length of the training symbol is about the same as length of time required to transmit the useful information and logic overhead.
The prior art methods accumulate multiple cells to form longer bursts diluting down the equalization overhead. The penalty for the use of longer bursts is proportionally longer quantizing, transfer and access delay. The penalty is yet greater in prior art systems that are fundamentally peer-to-peer asynchronous (no regular frame structure), because then the training symbol must be used as well for every down link transmission in addition to every up link transmission.
Theoretical Basis
The invention uses two steps which apart from each other are prior art. The first step is measuring the impulse response of the medium. The second step is defining the transmitted pulse shape which compensates the time dispersion of the medium to provide the original pulse shape with low distortion at the destination receiver.
Impulse Response Defined Transmission Medium
A valuable reference on this subject has been provided in "Filtering in the Time and Frequency Domain" by H. J. Blinchikov and A. I. Zverev. In this reference, the following paragraph appears:
"We now have reached the amazing conclusion that a linear system with constant parameters is uniquely characterized by the single function h(t), which is the system response to an impulse function applied at t=0. It is this function that is termed the system'impulse response to an arbitrary input by the way of the convolution integral." PA1 "Convolution and Deconvolution Convolving the impulse response of a digital filter with an input signal generates the corresponding filtered output. This document shows two ways of carrying out convolutions in MathCAD: . . . . The document also illustrates recovery of the original signal by deconvolution; you can carry out deconvolution efficiently in MathCAD by dividing the FFT of the output sequence by the FFT of the filter impulse response." "Electrical Engineering Application Pack--MathCAD," Copyright 1989 by MathSoft, Inc., One Kendall Square, Cambridge, Mass. 02139 (Page 93 of the Manual)
Barker Codes to Sound Medium and Observe Impulse Response
If the transmitted signal is a Barker code, the received signal is the impulse response of the medium with the degree of resolution and freedom from ambiguity limited by the code length. A 13-bit Barker will resolve paths arriving with one bit of time difference, and will lose accuracy from the ambiguity of response delayed longer than the 13-bit symbol. Longer codes will deal with larger amounts of time dispersion as a multiple of symbol length. The Barker is in effect a bandwidth limited pulse driver. The invention uses this information in the form of polar samples over the duration of the symbol. Averaging of repeated measurements decreases the uncertainty of these samples due to noise added in transmission. This technique is prior art.
Finite Impulse Response Pulse Shape Formation
There is art in generating the pulse shape representing the individual bits to obtain a defined and minimized frequency spectrum. To apply transform mathematics, this pulse shape must be mathematically and not empirically defined. A good pulse shape has been found to be the well known duo-binary pulse shape compressed to 2/3rds of its normal period. Each pulse, optimized for the distortionless medium, is given by equation 2! below. EQU T.sup.2 sin(.pi.t/T)/.pi.t(T-t) duobinary 1! EQU (2T/3).sup.2 sin(3.pi.t/2T)/.pi.t((2T/3)-t) compressed duobinary2!
The width of the main lobe of this pulse is spread over two bit intervals (as given by equation 2! above, rather than the three intervals of well-known duobinary pulses, as given by equation 1!). A short string of such pulses with polarity corresponding to a few different binary sequences results in a composite pattern as shown in FIG. 8 of Rypinski U.S. Pat. No. 5,388,126.
Consecutive pulses of the same sign add to a near constant dc level provided the detail of the low amplitude parts of the pulse extending a few bit intervals before and after the main lobe are taken into account. The capacity to define and generate a pulse shape extending over several bit intervals is what makes possible the use of this type of pulse shape which extends over several bit intervals and many of which must be overlaid in accordance with the data pattern. Modulations employing pulse shapes defined over plural bit intervals are referenced as "partial response" types.
The modification of this pulse shape to offset multipath time dispersion also must be made over a period of several bit intervals.
The use of Finite Impulse Response (FIR) filters to generate this and other pulse shapes is prior art. A convenient hardware method is based on shift register stepping at a sample rate which id a small integer multiple of the bit rate, where the pulse shape generated depends on the value of preceding and following bits as well as the current bits.
The impulse response is a measure of the medium. A new function must be generated which undoes the time and amplitude dispersion. This is similar to taking the pulse shown as an input and concentrating it back to the very narrow pulse that was used to generate the received shape.
Convolution and Deconvolution
These mathematical arts are described in the literature, where the functions are manipulations using the Fast Fourier Transform. The calculation of the needed wave shapes can be done with available computer applications, for example:
Station Transmit Pulse Shape
The desired transmit pulse shape is one which, after passing through the distortion introduced by the line as defined by the impulse response, produces the same pulse shape as was defined for the distortionless line.
The desired transmit pulse shape is calculated as one that a time convolution integral performed on the transmit pulse and the transmission medium impulse response produces the pulse shape defined for the distortion free medium.
This definition starts from the answer and works backward. The method of calculation uses the previously referenced deconvolution procedure.
Known Impulse Response Characteristics
In the last few years, a great deal of data has been collected on the propagation characteristics of microwave radio paths in buildings and on streets including measurements of quantities of impulse responses and their statistical evaluation. In particular NTIA-ITS (National Telecommunications and Information Administration, Institute for Telecommunication Sciences) has made high resolution polar measurements of impulse response at 1.5 Ghz in office and other environments.
The data presented includes samples of best and worst case responses, and statistical evaluation of large collections of samples vs. distance in several different environments. This kind of information is very important to the proportioning of the data rate and equalization symbol length.
This data is used to characterize the problem and its numeric proportions. The data is not used in any way in the parameters of the application of the invention.